Apply autoformat to the docs
Bug: None
Change-Id: Iea7823827d6f5a2d400ccbb208f91829165ee518
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220163
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34133}
diff --git a/pc/g3doc/dtls_transport.md b/pc/g3doc/dtls_transport.md
index 8b4ccaf..65206df 100644
--- a/pc/g3doc/dtls_transport.md
+++ b/pc/g3doc/dtls_transport.md
@@ -4,28 +4,45 @@
## Overview
WebRTC uses DTLS in two ways:
-* to negotiate keys for SRTP encryption using [DTLS-SRTP](https://www.rfc-editor.org/info/rfc5763)
-* as a transport for SCTP which is used by the Datachannel API
-The W3C WebRTC API represents this as the [DtlsTransport](https://w3c.github.io/webrtc-pc/#rtcdtlstransport-interface).
+* to negotiate keys for SRTP encryption using
+ [DTLS-SRTP](https://www.rfc-editor.org/info/rfc5763)
+* as a transport for SCTP which is used by the Datachannel API
-The DTLS handshake happens after the ICE transport becomes writable and has found a valid pair.
-It results in a set of keys being derived for DTLS-SRTP as well as a fingerprint of the remote certificate which is compared to the one given in the SDP `a=fingerprint:` line.
+The W3C WebRTC API represents this as the
+[DtlsTransport](https://w3c.github.io/webrtc-pc/#rtcdtlstransport-interface).
+
+The DTLS handshake happens after the ICE transport becomes writable and has
+found a valid pair. It results in a set of keys being derived for DTLS-SRTP as
+well as a fingerprint of the remote certificate which is compared to the one
+given in the SDP `a=fingerprint:` line.
This documentation provides an overview of how DTLS is implemented, i.e how the
following classes interact.
## webrtc::DtlsTransport
-The [`webrtc::DtlsTransport`][1] class is a wrapper around the `cricket::DtlsTransportInternal` and allows registering observers implementing the `webrtc::DtlsTransportObserverInterface`.
-The [`webrtc::DtlsTransportObserverInterface`][2] will provide updates to the observers, passing around a snapshot of the transports state such as the connection state, the remote certificate(s) and the SRTP ciphers as [`DtlsTransportInformation`][3].
-##cricket::DtlsTransportInternal
-The [`cricket::DtlsTransportInternal`][4] class is an interface. Its implementation is [`cricket::DtlsTransport`][5].
-The `cricket::DtlsTransport` sends and receives network packets via an ICE transport.
-It also demultiplexes DTLS packets and SRTP packets according to the scheme described in [RFC 5764](https://tools.ietf.org/html/rfc5764#section-5.1.2).
+The [`webrtc::DtlsTransport`][1] class is a wrapper around the
+`cricket::DtlsTransportInternal` and allows registering observers implementing
+the `webrtc::DtlsTransportObserverInterface`. The
+[`webrtc::DtlsTransportObserverInterface`][2] will provide updates to the
+observers, passing around a snapshot of the transports state such as the
+connection state, the remote certificate(s) and the SRTP ciphers as
+[`DtlsTransportInformation`][3].
+
+## cricket::DtlsTransportInternal
+
+The [`cricket::DtlsTransportInternal`][4] class is an interface. Its
+implementation is [`cricket::DtlsTransport`][5]. The `cricket::DtlsTransport`
+sends and receives network packets via an ICE transport. It also demultiplexes
+DTLS packets and SRTP packets according to the scheme described in
+[RFC 5764](https://tools.ietf.org/html/rfc5764#section-5.1.2).
## webrtc::DtlsSrtpTranport
-The [`webrtc::DtlsSrtpTransport`][6] class is responsіble for extracting the SRTP keys after the DTLS handshake as well as protection and unprotection of SRTP packets via its [`cricket::SrtpSession`][7].
+
+The [`webrtc::DtlsSrtpTransport`][6] class is responsіble for extracting the
+SRTP keys after the DTLS handshake as well as protection and unprotection of
+SRTP packets via its [`cricket::SrtpSession`][7].
[1]: https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/pc/dtls_transport.h;l=32;drc=6a55e7307b78edb50f94a1ff1ef8393d58218369
[2]: https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/api/dtls_transport_interface.h;l=76;drc=34437d5660a80393d631657329ef74c6538be25a
diff --git a/pc/g3doc/srtp.md b/pc/g3doc/srtp.md
index 3c806e1..4744615 100644
--- a/pc/g3doc/srtp.md
+++ b/pc/g3doc/srtp.md
@@ -2,36 +2,60 @@
<?% config.freshness.owner = 'hta' %?>
# SRTP in WebRTC
-WebRTC mandates encryption of media by means of the Secure Realtime Protocol, or SRTP, which is described in [RFC 3711](https://datatracker.ietf.org/doc/html/rfc3711).
-The key negotiation in WebRTC happens using DTLS-SRTP which is described in [RFC 5764](https://datatracker.ietf.org/doc/html/rfc5764).
-The older [SDES protocol](https://datatracker.ietf.org/doc/html/rfc4568) is implemented but not enabled by default.
+WebRTC mandates encryption of media by means of the Secure Realtime Protocol, or
+SRTP, which is described in
+[RFC 3711](https://datatracker.ietf.org/doc/html/rfc3711).
-Unencrypted RTP can be enabled for debugging purposes by setting the PeerConnections [`disable_encryption`][1] option to true.
+The key negotiation in WebRTC happens using DTLS-SRTP which is described in
+[RFC 5764](https://datatracker.ietf.org/doc/html/rfc5764). The older
+[SDES protocol](https://datatracker.ietf.org/doc/html/rfc4568) is implemented
+but not enabled by default.
+
+Unencrypted RTP can be enabled for debugging purposes by setting the
+PeerConnections [`disable_encryption`][1] option to true.
## Supported cipher suites
The implementation supports the following cipher suites:
-- SRTP_AES128_CM_HMAC_SHA1_80
-- SRTP_AEAD_AES_128_GCM
-- SRTP_AEAD_AES_256_GCM
-The SRTP_AES128_CM_HMAC_SHA1_32 cipher suite is accepted for audio-only connections if offered by the other side. It is not actively supported, see [SelectCrypto][2] for details.
+* SRTP_AES128_CM_HMAC_SHA1_80
+* SRTP_AEAD_AES_128_GCM
+* SRTP_AEAD_AES_256_GCM
-The cipher suite ordering allows a non-WebRTC peer to prefer GCM cipher suites, however they are not selected as default by two instances of the WebRTC library.
+The SRTP_AES128_CM_HMAC_SHA1_32 cipher suite is accepted for audio-only
+connections if offered by the other side. It is not actively supported, see
+[SelectCrypto][2] for details.
+
+The cipher suite ordering allows a non-WebRTC peer to prefer GCM cipher suites,
+however they are not selected as default by two instances of the WebRTC library.
## cricket::SrtpSession
-The [`cricket::SrtpSession`][3] is providing encryption and decryption of SRTP packets using [`libsrtp`](https://github.com/cisco/libsrtp). Keys will be provided by `SrtpTransport` or `DtlsSrtpTransport` in the [`SetSend`][4] and [`SetRecv`][5] methods.
-Encryption and decryption happens in-place in the [`ProtectRtp`][6], [`ProtectRtcp`][7], [`UnprotectRtp`][8] and [`UnprotectRtcp`][9] methods.
-The `SrtpSession` class also takes care of initializing and deinitializing `libsrtp` by keeping track of how many instances are being used.
+The [`cricket::SrtpSession`][3] is providing encryption and decryption of SRTP
+packets using [`libsrtp`](https://github.com/cisco/libsrtp). Keys will be
+provided by `SrtpTransport` or `DtlsSrtpTransport` in the [`SetSend`][4] and
+[`SetRecv`][5] methods.
+
+Encryption and decryption happens in-place in the [`ProtectRtp`][6],
+[`ProtectRtcp`][7], [`UnprotectRtp`][8] and [`UnprotectRtcp`][9] methods. The
+`SrtpSession` class also takes care of initializing and deinitializing `libsrtp`
+by keeping track of how many instances are being used.
## webrtc::SrtpTransport and webrtc::DtlsSrtpTransport
-The [`webrtc::SrtpTransport`][10] class is controlling the `SrtpSession` instances for RTP and RTCP. When [rtcp-mux](https://datatracker.ietf.org/doc/html/rfc5761) is used, the `SrtpSession` for RTCP is not needed.
-[`webrtc:DtlsSrtpTransport`][11] is a subclass of the `SrtpTransport` that extracts the keying material when the DTLS handshake is done and configures it in its base class. It will also become writable only once the DTLS handshake is done.
+The [`webrtc::SrtpTransport`][10] class is controlling the `SrtpSession`
+instances for RTP and RTCP. When
+[rtcp-mux](https://datatracker.ietf.org/doc/html/rfc5761) is used, the
+`SrtpSession` for RTCP is not needed.
+
+[`webrtc:DtlsSrtpTransport`][11] is a subclass of the `SrtpTransport` that
+extracts the keying material when the DTLS handshake is done and configures it
+in its base class. It will also become writable only once the DTLS handshake is
+done.
## cricket::SrtpFilter
+
The [`cricket::SrtpFilter`][12] class is used to negotiate SDES.
[1]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/peer_connection_interface.h;l=1413;drc=f467b445631189557d44de86a77ca6a0c3e2108d
